prestonlewis
Premium,MVM
join:2003-04-13
Sacramento, CA
 ·VoiceStick
 ·Comcast
 ·Pacific Bell - SBC
·DSL EXTREME
·Vonage
·VoicePulse
| Yahoo Voice: Registered but no call can go out
OK. After a lot of trial and error. I'm registered at Yahoo Voice. It even tells me how much money I have on my account on the SPA-3102 page. However, no calls can go out. I just get a fast busy. Could someone take a look at my SPA pages and tell me what I'm doing wrong?
By the way, my SPA3102 had version 3 firmware (no auth or turn servers). Had to upgrade to version 5.1.7 firmware to get those servers listed.
I did try changing my dial string to 10 digit dialing only but still got a fast busy. Also note you don't put the at yahoo.com, only your Yahoo user name. |
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Stewart
join:2005-07-13
edit:
May 8th, @09:40PM
| Sorry, I can't spot the error. 10-digit dialing definitely won't work, unless you're calling e.g. Bangkok, because Yahoo expects the number to start with the country code.
I see that you have SIP Debug set to full. What error do you get? Immediate or after retry with Authorization header? Does the INVITE look correct? Sending plausible SDP?
I'm running 5.2.3, but don't know if that makes a difference. (5.2.3 no longer has a Yahoo! line on the Info page.)
My codec is set to G.711u only (yes, I know that Messenger uses G.729).
There are some differences in NAT mapping -- if it's easy for you to try, put it on a public IP and set NAT Mapping Enable to no.
Also, I have RTP Packet Size: 0.020
Edit: also noticed that SIP Remote-Party-ID: no and
Use Auth ID: no |
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mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
| reply to prestonlewis
said by prestonlewis  :I just get a fast busy. Usually, a fast busy signal is an indication of a dial-plan string issue. |
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hwittenb
join:2003-12-20
| reply to prestonlewis
I had Stewart's Yahoo deal working yesterday with my SPA3102 which is running firmware 5.1.7(GW). I left it registered and today it was still registered but a call got the same result as you got ... fast busy. A sip debug trace showed that the Yahoo server was returning 403 Forbidden. A Google search says 403 Forbidden means various things but seems to be an acknowledgment that the server got the request but won't provide the service.
I tried my Yahoo softphone on my computer and it still worked.
I thought maybe I would see what happens if I power cycle the SPA3102. That was a mistake. The SPA wouldn't reboot, kept rebooting with the sip debug trace information showing some weird error revolving around the Yahoo code.
The solution was to pull the ethernet cable, reboot and then use the IVR and reset the SPA to factory specs, a major nosebleed.
I think I'll write off the Yahoo excursion as a bad dream. |
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Stewart
join:2005-07-13
| reply to prestonlewis
The SPA-2102 does not play well with Messenger. Yahoo permits only one registration per account; when Messenger registers, it gets yanked from the SPA, which will get a 403 if you try to call. However, the SPA doesn't attempt to get it back; it just keeps showing 'registered'. The only way I know to recover is to exit Messenger (be sure that you don't just let it minimize to the taskbar notification area), then reboot the SPA so it registers again.
hwittenb, sorry to hear about your trouble. IMO, the reset was overkill. If you had e.g. unplugged the Internet port and plugged a PC into the Ethernet (LAN) port, you almost certainly would have been able to access the web interface to change config, upgrade firmware, etc. I've rebooted the SPA perhaps 50 times (though only a few with a Yahoo config) with no trouble. |
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hwittenb
join:2003-12-20
| said by Stewart :hwittenb, sorry to hear about your trouble. Mark Twain once wrote “The cat, having sat upon a hot stove lid, will not sit upon a hot stove lid again. But he won't sit upon a cold stove lid, either.”
I think I will just let my balance of 6 Singapore Dollars on Yahoo Voice just ride.
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Pinan
Hypnotic Tweaker
ExMod 2000-03
join:2000-09-02
Murrieta, CA | Give it another go. Stewart set me up on Yahoo and it works great on my SPA2102. |
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goodchefro
join:2007-02-21
Macomb, IL
·Logonix Corporation
| Stewart,
thanks much! for the various tips offered in both threads.
I got it to work. The single "bug" right now is to figure out why it doesn't ring. I dialed three different toll-free numbers and while the calls completed successfully, none rang.
I have the "DTMF Relay MIME Type" field as RFC 2833, instead the default "application/dtmf-relay". Could that be the culprit?
Thanks again. 
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Stewart
join:2005-07-13
| If the unit is behind a NAT and you are not using TURN server, you must forward the UDP port range specified by RTP Port Min to RTP Port Max to the SPA. That's because the SPA fails to send RTP during 183 Progress (IMO a bug; I have a ticket open with Linksys on this issue). |
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goodchefro
join:2007-02-21
Macomb, IL
·Logonix Corporation
edit:
May 9th, @11:34PM
| Stewart,
I'm behind a NAT. As a matter of fact, the UDP port range specified by RTP Port Min and Max actually is already forwarded.
Before changing the TURN server from "0.0.0.0." to "relay.voice.yahoo.com" I had dial tone, but the calls would not complete. Only after changing to the latter the calls started to go through.
(Edit: forgot here to mention that I also added my external IP in the SIP tab)
As far as ringing goes, it makes no difference what I use in the DTMF Relay field (as I was asking earlier this morning) since neither the default value, nor RFC 2833 alleviated the no ring occurence.
I even changed the "Sticky 183" field to "yes", but to no avail.
I'll keep playing with it, maybe something different will do the trick.
Thanks. |
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prestonlewis
Premium,MVM
join:2003-04-13
Sacramento, CA
·VoiceStick
·Comcast
·Pacific Bell - SBC
·DSL EXTREME
·Vonage
·VoicePulse
| reply to prestonlewis
Stewart, since you seem to be the Yahoo Voice God here in the forums, could you post a pic of your SIP/Line 1 pages here for us lowly unworthy minions?
For those who don't know how to take a pic of an entire web page when part of it requires scrolling down to see:
Use Firefox. Go to the Tools, then addon tab. Find an addon named "Snapshot" and install it in your Firefox. Reboot Firefox. You'll see a big S surrounded by red in the top right. Click the little triangle to the right. You'll get the option of taking a "snapshot" of just the visible page or the entire page including the invisible part that requires scrolling to see. Save as a JPG, not the default png (unless you know all about png, which I don't). The start a reply in this thread and upload the two pics (SIP/Line1) The entire page should be visible from top to bottom. Snapshot works great for this kind of thing. |
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goodchefro
join:2007-02-21
Macomb, IL
·Logonix Corporation
| SIP TAB |
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goodchefro
join:2007-02-21
Macomb, IL
·Logonix Corporation
| LINE 2 - CONFIGURED WITH YAHOO
IT WORKS (except for the lack of ringing) |
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Stewart
join:2005-07-13 | reply to prestonlewis
Both of you should use SIP Debug and Wireshark to see what is going wrong. Note that setting TURN Server to "relay.voice.yahoo.com" will not work; you must use "relay.voice.yahoo.com:443" . |
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goodchefro
join:2007-02-21
Macomb, IL
·Logonix Corporation
| said by Stewart : you must use "relay.voice.yahoo.com:443" That's what I have; the screenshot didn't fit the :443 port in the image and I didn't realize that; I posted the screenshot for Lewis.
By the way: if I turn the SIP Debug feature in the ATA STOPS responding, with the power LED blinking mostly red... |
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Stewart
join:2005-07-13
| reply to prestonlewis
I have no idea why SIP Debug doesn't work for you. I have: Syslog Server: (empty); Debug Server: (static private address of PC); Debug Level: 0; SIP Debug Option: full. Running firmware v5.2.3 . Using Wireshark to capture the pseudo-syslog packets and display them. Tip: if you tell Wireshark to decode destination port 514 as SIP, the SIP will be properly decoded for display.
If you can't get this to work, rig some means of capturing the SIP directly (dumb hub, managed switch, PC with two NICs, etc.) |
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Pinan
Hypnotic Tweaker
ExMod 2000-03
join:2000-09-02
Murrieta, CA | reply to goodchefro
FWIW. I have Nat Mapping Enable set to Yes
and EXT SIP Port: blank.
Also, TURN Server:0.0.0.0
I do not have a DID, only outgoing. |
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